For this, it works randomly. I have decided to work on SIP and forgot about h323 with Asterisk. I spent nights and nights trying to figure out how it works, but decided to move on.
Now we are running SIP, things are better. Selon Chetan Sarva <[EMAIL PROTECTED]>: > Did you ever find a solution to this problem? > > [EMAIL PROTECTED] wrote: > > >All, > > > >I have tried very hard to make asterisk work with h323 but still strying: > > > >I have been successful making this work > > > >SIP --> Asterisk --> h323 --> termination ; > > > >But the following: > > > >h323 --> asterisk --> h323 --> Termination : works , call set up is ok but > then > >no audio is applied .There is no NAT here at all are public. > > > >I also tried > > > >h323 --> asterisk --> SIP --> terminatino: I have same problem here, audio > > > >I use g723 codec (passthrough ) > > > >Can anyone advise what is to look or is it meant not to work anyway ? > > > > > > > > _______________________________________________ > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
